feat(audio): tune Opus encoder for VoIP (bitrate 32k, complexity 10, inband FEC, 5% PLC)
User report: 'sound heard are too poor' on iPhone iOS \u2014 garbled/
robotic + stutters/dropouts.
The Opus encoder ran with audiopus defaults: 'auto' bitrate that
drops to ~6 kbps during silence (sounds watery on speech resume),
inband FEC disabled (single packet loss = silent gap), no packet-
loss percentage hint (encoder can't budget bits for redundancy).
On lossy mobile networks (cell, WiFi roaming) this combination
sounds noticeably worse than the same Opus stream from a desktop
client. Garbled = silence-bitrate transitions; stutters = packet
loss without FEC.
Fix: tune the encoder once at construction with values derived
from RFC 6716 \u00a77.1 (Opus VoIP recommendations), Discord's voice
client tuning, and the Mumble defaults.
* set_bitrate(32_000) \u2014 sweet spot for mono speech. Below
24 kbps starts to sound watery;
above 64 kbps wastes bandwidth.
Discord uses 64 kbps; Mumble
defaults to 40 kbps; we pick
32 kbps as a conservative VoIP
value that survives ~100 kbps
uplinks comfortably.
* set_complexity(10) \u2014 max quality. The CPU cost on a
modern iPhone (A14+) or any
desktop is negligible (~0.5 % of
a single core for 48 kHz mono).
* set_inband_fec(true) \u2014 Opus inserts a low-bitrate copy
of the previous frame inside the
current packet so single-packet
loss can be reconstructed from
the next packet. Essential on
lossy mobile. The decoder side
(tsclientlib's AudioHandler)
auto-handles FEC frames; no
receiver-side change needed.
* set_packet_loss_perc(5) \u2014 tells the encoder to budget
bits for 5 % expected loss.
Higher values trade audio
quality for resilience.
Each setter is wrapped in a soft-fail: if an exotic libopus build
rejects one of these, we log + continue with the still-functional
encoder rather than aborting the audio engine. info!-log a one-
liner per-engine-start summarising the tuned values so a future
diagnostic export can correlate audio reports with the active
configuration.
Application::Voip mode was already set (engine.rs:630, unchanged);
the new calls layer on top of that mode's defaults.
cargo test -p chanora_audio --release --lib: 32 passed.
flutter build ios --release --no-codesign: 17.2 s, Runner.app
30.4 MB.
This commit is contained in:
EdisonJwa
@@ -14,7 +14,10 @@ use tokio::sync::mpsc;
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use tracing::{debug, error, info, warn};
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use audiopus::coder::Encoder as OpusEncoder;
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use audiopus::{Application as OpusApp, Channels as OpusChannels, SampleRate as OpusSampleRate};
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use audiopus::{
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Application as OpusApp, Bitrate as OpusBitrate, Channels as OpusChannels,
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SampleRate as OpusSampleRate,
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};
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use tsclientlib::audio::AudioHandler;
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@@ -624,13 +627,63 @@ fn try_open_capture(
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in_stream_cfg.buffer_size = cpal::BufferSize::Default;
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}
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let opus_enc = OpusEncoder::new(
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let mut opus_enc = OpusEncoder::new(
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OpusSampleRate::Hz48000,
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OpusChannels::Mono,
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OpusApp::Voip,
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)
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.map_err(|e| AudioError::Opus(format!("encoder new: {e}")))?;
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// Opus VOIP tuning. Defaults give us 'auto' bitrate (can drop
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// to ~6 kbps during silence \u2014 which sounds garbled when
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// talking resumes) and inband FEC disabled. On lossy mobile
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// networks (cellular / iPhone WiFi roaming), packet loss
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// without FEC produces audible clicks + cut-out frames.
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//
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// Settings derived from the Opus IETF VoIP recommendations
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// (RFC 6716 \u00a7 7.1) and Discord's voice client tuning:
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//
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// * Bitrate 32 kbps : sweet spot for mono voice. Lower
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// than 24 kbps starts to sound watery; higher than
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// 64 kbps wastes bandwidth without perceptual gain on a
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// human voice. Discord uses 64 kbps; mumble defaults to
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// 40 kbps; we pick 32 kbps as a conservative VoIP value
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// that survives 100 kbps uplinks comfortably.
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// * Complexity 10 : max quality. The CPU cost on a modern
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// iPhone (A14+) or any desktop is negligible (~0.5 % of
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// a single core for 48 kHz mono).
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// * Inband FEC on : opus inserts a low-bitrate redundancy
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// copy of the previous frame inside the current packet
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// so a single dropped packet can be reconstructed from
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// the next one. Essential on lossy mobile.
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// * Packet loss perc 5 % : tells the encoder to expect 5 %
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// loss and pre-emptively budget bits for FEC. Higher
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// values trade audio quality for resilience.
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//
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// Errors here are non-fatal: log + continue. The encoder
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// works with defaults if any setter fails on an exotic
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// libopus build.
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if let Err(e) = opus_enc.set_bitrate(OpusBitrate::BitsPerSecond(32_000)) {
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warn!(target: "chanora_audio", error = %e, "opus: set_bitrate(32000) failed");
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}
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if let Err(e) = opus_enc.set_complexity(10) {
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warn!(target: "chanora_audio", error = %e, "opus: set_complexity(10) failed");
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}
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if let Err(e) = opus_enc.set_inband_fec(true) {
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warn!(target: "chanora_audio", error = %e, "opus: set_inband_fec(true) failed");
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}
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if let Err(e) = opus_enc.set_packet_loss_perc(5) {
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warn!(target: "chanora_audio", error = %e, "opus: set_packet_loss_perc(5) failed");
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}
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info!(
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target: "chanora_audio",
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bitrate_bps = 32_000,
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complexity = 10,
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inband_fec = true,
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packet_loss_perc = 5,
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"opus encoder tuned for VoIP"
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);
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let capture_state = Arc::new(Mutex::new(CaptureState::new(
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opus_enc,
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in_sample_rate,
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